[..] |
ASTERISK-18000: If asterisk starts before my Internet connection is up, asterisk cannot register to any of my registrations and reports DNS errors |
ASTERISK-18002: Voicemail MWI no longer is sent |
ASTERISK-18019: MWI Subscriptions have no effect and are not handled correctly |
ASTERISK-18023: Typo in Goto/GotoIf documentation |
ASTERISK-18024: Incorrect data in CDRs (billsec >> durration) |
ASTERISK-18025: CPU spikes when using timerfd timing |
ASTERISK-18026: deadlock every 2 hours |
ASTERISK-18027: SMS queued with smsq not sent |
ASTERISK-18028: Asterisk 1.8 Realtime provider |
ASTERISK-18029: Aastra 480i does not work with Asterisk 1.8 |
ASTERISK-18031: Deadlock in ast_async_goto() because of wrong locking order |
ASTERISK-18032: [patch] - IPv6 and IPv4 NAT not working |
ASTERISK-18036: Enhance Automated Dialplan Tests |
ASTERISK-18037: [patch] Add hold status with CEL and setting a channel var HOLDING via ast_moh_stop/start and potential res_musiconhold fixup. |
ASTERISK-18038: Only one digit on call transfer |
ASTERISK-18039: Realtime music restarts from beginning each time |
ASTERISK-18040: Asterisk segfaults on shutdown when confbridge is in use |
ASTERISK-18041: distribute voicemail to other boxes using database lookup |
ASTERISK-18042: [patch] information from unsolicited MWI is lost during asterisk restart; cache to ast_db to avoid this |
ASTERISK-18043: MeetMe Not Respecting SetMusicOnHold() |
ASTERISK-18044: extenpatternmatchnew breaks ability for caller to exit queue |
ASTERISK-18045: gtalk should subscribe to res_stun_monitor |
ASTERISK-18046: commit code for 'stun show status' |
ASTERISK-18054: Huge list of frozen SIP channels |
ASTERISK-18055: Attempting to register to remote server that returns 404 causes no further registration attempts |
ASTERISK-18056: Test Issue - Just Ignore |
ASTERISK-18058: OOH323 Fails to Identify Dead Peer and Rings Indeffinitely |
ASTERISK-18060: Playback in h exten triggers a null pointer in sip_setoptions resulting in a crash |
ASTERISK-18061: Asterisk periodically locksup, using CPU time. |
ASTERISK-18062: menuselect dependency resolution is broken with gcc 4.6 |
ASTERISK-18063: Flooding with [Jun 24 19:33:17] WARNING[6995]: chan_sip.c:6213 sip_write: Asked to transmit frame type g726, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw) |
ASTERISK-18064: Asterisk's AMI over HTTP adds an extra \r\n in the headers causing strict parsers to fail |
ASTERISK-18065: 1.4 fails to build with gcc 4.6 in dev-mode: chan_phone.c:307:9: error: the comparison will always evaluate as 'true' |
ASTERISK-18066: Attended transfert with sendrpid=yes and directedmedia=yes with aastra phone, return 500 error and not works |
ASTERISK-18067: REGRESSION: After upgrading from 1.4.41 to *1.8.4.2 a SIP extension with a voicemail box can no longer monitor mwi of another extension |
ASTERISK-18068: Request to use ConfBridge() instead of MeetMe() in Page() |
ASTERISK-18069: [patch] app_queue Add Login Time and Last Paused Times to Queue Members |
ASTERISK-18070: Asterisk 1.4.22 - 1.4.35 crashes once or twice a week |
ASTERISK-18071: app_queue Add Member StateInterface to Output of "queue show" (CLI) and "QueueStatus" (AMI) |
ASTERISK-18072: Segfault when using Directory() |
ASTERISK-18073: If INVITE transaction runs in parallel with INFO transaction, 200 OK for INVITE does not contain contact headers |
ASTERISK-18074: SIP messages stop being processed |
ASTERISK-18076: Asterisk 1.6.1 won't "answer" the phone when using a callcentric sip trunk |
ASTERISK-18077: When in queue on g722 with interruptions, music on hold can get stuck and no longer play |
ASTERISK-18078: [patch] Segfault when publishing device states via XMPP and not connected |
ASTERISK-18079: Flash doesn't break dialtone on DAHDI FXS channels |
ASTERISK-18080: DUNDi lookups cause deadlocks |
ASTERISK-18081: Queues - Missed calls - SIP cancel reason |
ASTERISK-18082: Deadlock of SIP or segfault when doing REFERs |
ASTERISK-18083: "r" dial params stop give ringback if M macro used |
ASTERISK-18084: chan_gtalk only receives calls from Gtalk Android but not viceversa |
ASTERISK-18085: Asterisk crashes when using incorrect Dial() syntax |
ASTERISK-18086: Can't place calls on hold with certain IP phones (aastra 9133i and sipdroid soft phone) |
ASTERISK-18087: asterisk will crash on "module reload" |
ASTERISK-18089: Using comment ;--- causes dialplan corruption |
ASTERISK-18090: ERROR[15785]: chan_iax2.c:1742 iax2_getpeername: Bad address cast to IPv4 |
ASTERISK-18091: Every "sip notify" cmd open a udptl port, and does not free it |
ASTERISK-18092: asterisk segfault libpthread-2.9.so |
ASTERISK-18093: Add an extra argument to app_originate (dialplan originate()) to state that it should run async |
ASTERISK-18094: iLBC (30ms packet) to G711 (20ms) ULAW transcoding sounds "robotic" |
ASTERISK-18095: Misleading "Call from 'xxx' to extension 'yyy' rejected because extension not found in context" due to missing SIP domain |
ASTERISK-18096: While receiving an AMI event:MeetmeLeave, the Uservalue: has non-ascii values |
ASTERISK-18097: svn revision 326678 of the Asterisk Trunk fails to compile on SQLITE3 error |
ASTERISK-18100: chan_sip not responding sometimes until asterisk restart |
ASTERISK-18101: Asterisk 1.8 Deadlock in app_queue |
ASTERISK-18102: Asterisk does not start when compiled and loaded in a Cygwin environment |
ASTERISK-18103: asterisk 1.6.2.19 core dump on reload |
ASTERISK-18104: asterisk terminates on module load attempt |
ASTERISK-18105: most of asterisk modules are unbuildable in cygwin environment |
ASTERISK-18106: specifying features.conf to be loaded by static realtime is reported as bound when extconfig.conf is parsed but is ignored |
ASTERISK-18107: Asterisk is not answering correctly the OPTIONS messages sent by SIP provider. |
ASTERISK-18108: building asterisk under cygwin: errors in main/editline/np/unvis.c |
ASTERISK-18109: Segfault in shell_helper in func_shell.c |
ASTERISK-18111: chan_iax2 module unbuildable in cygwin environment |
ASTERISK-18127: Queues - Missed calls - SIP cancel reason |
ASTERISK-18128: It started to crash several times x day 3 to 4 days ago |
ASTERISK-18129: Urgent need for IP permit-deny at the GLOBAL level, not at the peer level |
ASTERISK-18130: core show application sendfax/receivefax shows XML docs for app_fax version even when res_fax is loaded |
ASTERISK-18131: IAX_COMMAND_HANGUP-Frame not sent after Hangup(${HANGUPCAUSE}) |
ASTERISK-18132: Asterisk segmentation fault (core dump) |
ASTERISK-18133: AsteriskGUI consistently crashes Asterisk |
ASTERISK-18134: Reload hangs when using func_odbc for hints. |
ASTERISK-18135: removal of a specific extension that happens to be a prefix of another extension causes memory corruption |
ASTERISK-18136: r319652 causes deadlock with REFER |
ASTERISK-18137: Configurations saved in ~/.asterisk.makeopts ignored |
ASTERISK-18138: lock.c:280 __ast_pthread_mutex_lock: ccss.c line 3478 (ast_cc_offer): Error obtaining mutex: Invalid argument |
ASTERISK-18139: Change libsrtp "not found" message to not suggest --prefix be used |
ASTERISK-18140: Expires handling in SUBSCRIBE confuses the absence of the Expires header field with an unsubscribe action. |
ASTERISK-18141: "make menuselect" in 1.8.5.0 does not alphabetically sort options |
ASTERISK-18142: unresponsive to sip requests |
ASTERISK-18143: [patch] Add diversion header to 302 redirect SIP message if we have redirection data |
ASTERISK-18144: PickupChan not working correctly |
ASTERISK-18145: [regression] With IMAP Voicemail MWI not cleared or set for SIP device |
ASTERISK-18146: Unable to register an endpoint after upgrading from 1.4.2.20 to 1.8.3.1 |
ASTERISK-18147: SIP Phone (Cisco)-> Asterisk 1.4.31 -> IAX-> Asterisk 1.6.2.19 > SIP provider -> PSTN IVR DTMF - DTMF broken after upgrade 1.4 to 1.6 or 1.8 |
ASTERISK-18149: Dead lock in parking chan_sip handle_request_refer |
ASTERISK-18151: Crash in Asterisk under 64 bit arch after authenticating correctly via Asterisk GUI |
ASTERISK-18152: [patch] AMI AgentCalled CallerID is Agent extension/name |
ASTERISK-18153: Wrong Caller ID Sent in Invite From Header From Realtime SIP User |
ASTERISK-18154: Regression: Channel Restarts, Congestion, Limited Call Capacity on PRI |
ASTERISK-18155: bridge_softmix.c line 149 (softmix_bridge_leave): Error: attempt to destroy invalid mutex '&sc->lock' |
ASTERISK-18156: SIP messages stop being processed with res_timing_dahdi |
ASTERISK-18160: Delay between sip unregistration and peer status change |
ASTERISK-18161: res_fax.conf: crash if invalid |
ASTERISK-18162: Asterisk manager (AMI) Redirect = No CDR written |
ASTERISK-18163: DeadLock in asterisk manager interface |
ASTERISK-18164: MixMonitor - Handle is not released before HangUp |
ASTERISK-18165: sms sending does not work |
ASTERISK-18166: Deadlock: asterisk isn't responding to any sip package anymore |
ASTERISK-18167: Functions and modules fail to load |
ASTERISK-18168: WIkibot not respecting argsep parameter in parameter tags. |
ASTERISK-18169: Asterisk crashes on incoming call |
ASTERISK-18170: Asterisk 1.8 bri cards (b410p or openvox ) libpri |
ASTERISK-18171: Ajax Bug Asterisk 1.8 Manager using mxml |
ASTERISK-18172: SendDTMF with duration |
ASTERISK-18173: Asterisk sip stack crashing |
ASTERISK-18174: Couldn't execute statment: SQL logic error or missing database |
ASTERISK-18193: Asterisk doesnt work |
ASTERISK-18194: Asterisk 1.6. No audio when using MixMonitor on sip channels |
ASTERISK-18195: CLI output is limited to 24000 bytes, running 'dialplan reload' in verbose mode |
ASTERISK-18196: Asterisk high cpu usage |
ASTERISK-18197: Deadlock - Periodic Deadlock on Transferring Calls |
ASTERISK-18199: [regression] LIMIT_CONNECT_FILE does not get played to caller when using dial() app |
ASTERISK-18200: CLI output is limited to 24000 bytes, running 'dialplan reload' in verbose mode |
ASTERISK-18201: Asterisk should fall back to AVP when SRTP module is not loaded and both SAVP and AVP have been offered |
ASTERISK-18202: deadlock in sip |
ASTERISK-18203: Problems with NAT on realtime peers (and maybe static ones) |
ASTERISK-18204: Mute All Participants |
ASTERISK-18205: Deadlock in app_queue when loading real-time queues and handling state change. |
ASTERISK-18206: CLONE - chan_gtalk only receives calls from Gtalk Android |
ASTERISK-18207: externnotify script called with (null) context parameter during pollmessages run, essentially stopping it from running. |
ASTERISK-18208: logger doesn't append data to queue_log file after asterisk starts |
ASTERISK-18209: Dialing an incomplete number via dahdi causes a timeout |
ASTERISK-18210: chan_iax2 Asterisk crashes after client "crash" |
ASTERISK-18211: Asterisk deadlock |
ASTERISK-18212: [patch] German saydigits algorithm inserts spurious "and" |
ASTERISK-18213: asterisk (1.6.2.19 and 10.0.0-beta1) don't compile on OSX 10.7 (Lion) |
ASTERISK-18214: MixMonitor not parsing string variables |
ASTERISK-18215: asterisk stop response on SIP messages |
ASTERISK-18216: Global variables not set after restart. Must reload everytime. |
ASTERISK-18217: [request] Easy generation of AOC (advice of charge) messages |
ASTERISK-18218: 10beta1 ooh323 outbound call doesn't work |
ASTERISK-18219: t38 gateway doesn't work with ooh323 |
ASTERISK-18220: MixMonitor stops recording during attended Transfer |
ASTERISK-18221: Connecting to my prosody xmpp server crashes res_jabber.so with pkt->from = NULL |
ASTERISK-18222: Pickupchan of a local channel segfaults if 2 users pickup at same time |
ASTERISK-18223: using http manager commands causes global file descriptor instability, crashing Asterisk |
ASTERISK-18224: CDR(accountcode) not accessable to 'Local' channels |
ASTERISK-18225: SIP channels are getting stuck after picking up calls |
ASTERISK-18226: ConfBridge with two participants passes hold state to the other party. If a 3rd one joins that party stays deaf |
ASTERISK-18227: Manual module loading (with global symbols) broken since 1.4.28 |
ASTERISK-18228: dial() app option r playback early media |
ASTERISK-18230: sometimes dialplan switches disappear when merging contexts between pbx_lua and pbx_config |
ASTERISK-18231: asterisk crashes when doing multiple ConfBridge |
ASTERISK-18232: Broken REGISTER sent to IPv4 server when bindaddr=[::] |
ASTERISK-18233: Asterisk does not continue AGI code execution after the channel is hungup by caller. |
ASTERISK-18234: IAX2 wont start ring back |
ASTERISK-18235: IAX2 wont start ring back early media |
ASTERISK-18236: Segmentation fault after second chan_h323 unload |
ASTERISK-18237: chan_h323 fails to determine IP/resolve host if host parameter is IP |
ASTERISK-18238: local channel doesn't use language of requestor channel |
ASTERISK-18239: crash in 1.8 |
ASTERISK-18240: UniqueID is posted twice to CDR when CDR(userfield) is extended/broken in the process. |
ASTERISK-18243: VoiceMail application fails to assign some DTMF codes for application exit when using d() option with context |
ASTERISK-18244: When caller hangs up on AGI dial app the code does not complete execution |
ASTERISK-18245: Forwarding an Urgent voicemail through VoiceMail to a mailbox that has not been created fails |
ASTERISK-18246: When leaving a voicemail marked as Urgent with forwarding recipients, the forward_urgent_auto flag is not respected |
ASTERISK-18247: Asterisk taking up 673 mb of ram in idle state |
ASTERISK-18248: SIP crash but Asterisk stay UP |
ASTERISK-18249: Deadlocks when using a switch statement |
ASTERISK-18250: Resource leak in timerfd |
ASTERISK-18251: ConfBridge CLI Output Not As Clean As MeetMe |
ASTERISK-18252: queue_log mysql time column data format |
ASTERISK-18253: Init service isn't compatible with LSB |
ASTERISK-18254: say_digits() app sound distortion |
ASTERISK-18255: [regression] Non-portable SQL added to app_voicemail |
ASTERISK-18257: Develop automated test for SQL portability in Asterisk |
ASTERISK-18259: Asterisk 1.8.5.0 not putting p-asserted into ringing or answered SIP messages |
ASTERISK-18261: CLONE -asterisk does not starts |
ASTERISK-18263: sip directmedia nonat handling / unreachable code |
ASTERISK-18264: [patch] Generate security events in chan_sip using new Security Events Framework |
ASTERISK-18265: CLONE - [patch] Memory Leak in app_queue |
ASTERISK-18266: it should be posible to authenticate sip devices using name different than the section header |
ASTERISK-18267: Channel locks after 'core restart now' and 'core reload' |
ASTERISK-18268: Improve Menuselect and Asterisk CLI for Module Support Status and CLI |
ASTERISK-18269: Calls getting stuck when dialing *8 |
ASTERISK-18270: Useless message pops every time there is a bridging |
ASTERISK-18271: Pattern matching with res_config_mysql extensions does not behave as expected |
ASTERISK-18272: Segfault when using HTTP Digest Auth when accessing /amanager |
ASTERISK-18273: Orphaned channels after pickup |
ASTERISK-18275: DTMF blind transfer continues in dialplan after transfer. |
ASTERISK-18277: Memory leak in chan_sip / realtime_peer() / mysql |
ASTERISK-18278: cdr_adaptative_odbc does not write CHAR fields |
ASTERISK-18279: on-demand recording (*1) filename not being generated correctly due to incorrect uniqueid in filename |
ASTERISK-18280: astcanary does not get started when asterisk is startetd with -U <user> |
ASTERISK-18281: ERROR[3385]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x0 for 0x11f76860 |
ASTERISK-18282: Voicemail will not honor forcename / forcegreetings if password is changed and user hangs up |
ASTERISK-18283: Call Completion CallCompletionRequest No Core Instance |
ASTERISK-18284: Queue remove member command will not accept arguments diplayed by tab complete |
ASTERISK-18285: Call Completion CallCompletionRequest No Core Instance WHEN call-limit option set to 1 in sip.conf |
ASTERISK-18286: [patch] 'Silence' is truncated in Record() |
ASTERISK-18287: Processing in sig_pri.c corrupts memory |
ASTERISK-18288: peer->mvipvt needs locking |
ASTERISK-18289: FreeBSD: AsteriskUnitTests /main/netsock2/parsing failing |
ASTERISK-18290: Mac OSX: Fails to install Asterisk |
ASTERISK-18291: Sequence number roll over in res_rtp_multicast.c |
ASTERISK-18292: Release Asterisk 1.8.6.0-rc2 |
ASTERISK-18293: Merge: ASTERISK 18109 |
ASTERISK-18294: Merge: ASTERISK 18290 |
ASTERISK-18295: Merge: ASTERISK 18289 |
ASTERISK-18297: Realtime Asterisk Voice Mail - Sendvoicemail=yes option |
ASTERISK-18298: (Call Completion / SIP) SUBSCRIBE fails using TLS |
ASTERISK-18299: Merge: ASTERISK 18082 |
ASTERISK-18300: Merge: ASTERISK 18166 |
ASTERISK-18301: Outgoing calls fail in chan_gtalk |
ASTERISK-18302: System Deadlock, No calls inbound or outbound |
ASTERISK-18303: Problem with batch-creation of astdb entries |
ASTERISK-18304: Problem with batch-creation of astdb entries |
ASTERISK-18305: Problem with batch-creation of astdb entries |
ASTERISK-18306: Problem with batch-creation of astdb entries |
ASTERISK-18307: Problem with batch-creation of astdb entries |
ASTERISK-18308: Problem with batch-creation of astdb entries |
ASTERISK-18309: Termination of active call after 20 seconds, because of "Maximum retries exceeded on transmission" |
ASTERISK-18310: Termination of active call because of "Maximum retries exceeded on transmission" |
ASTERISK-18311: Termination of active call because of Maximum retries exceeded on transmission |
ASTERISK-18312: Termination of active call |
ASTERISK-18313: termination_of_active_call |
ASTERISK-18314: termination_of_active_call |
ASTERISK-18315: Problem with batch-creation of astdb entries |
ASTERISK-18316: Merge: ASTERISK 18301 |
ASTERISK-18317: Locking problems with unloading/loading chan_dahdi |
ASTERISK-18318: Suspect deadlock in res_ais |
ASTERISK-18319: [patch] Optimize chan_sip.c check_rtp_timeout() function |
ASTERISK-18320: Outgoing calls fail in chan_gtalk redux |
ASTERISK-18321: dynamic_exclude_static option with (temporary) unreachable DNS cause the abend |
ASTERISK-18322: ooh323 , alternate gatekeeper |
ASTERISK-18323: Memory leak in lock.h |
ASTERISK-18324: Kill the last user of a meetme at exit |
ASTERISK-18325: "Asked to transmit frame type" slows down all the calls |
ASTERISK-18326: Bugs in mISDNuser for NT-PTMP mode causes loss of avaliable procids |
ASTERISK-18327: [patch] Monitoring own ip with res_stun_monitor fails when local ip changes |
ASTERISK-18328: Asterisk locked |
ASTERISK-18330: IAX2 trunk audio problems |
ASTERISK-18331: app_sms failure |
ASTERISK-18332: chan_dahdi does not seem to report User Rate from Bearer Capabiities |
ASTERISK-18333: Chanspy g() option seem to now need precise SPYGROUP name |
ASTERISK-18334: chan_dahdi doesn't reset B-channels after calls. |
ASTERISK-18335: configure fails if there's a space in the current dir |
ASTERISK-18336: chan_vpb: build warnings with gcc 4.6 |
ASTERISK-18337: chan_h323: warnings when built with gcc 4.6 |
ASTERISK-18338: DESTDIR does not work when directory has spaces |
ASTERISK-18339: Regression: Loss of DTMF signals through asterisk |
ASTERISK-18340: CLONE - directmedia or reinvite not working when calling extension that's located an a different asterisk node |
ASTERISK-18341: REGEX function documentation |
ASTERISK-18342: close() before SSL_Shutdown() in ssl_close() |
ASTERISK-18343: extenpatternmatchnew fails to correctly respect dialplan order (regex match before exact in an included context) |
ASTERISK-18344: The SIP message "... rejected because extension not found in context ..." lacks vital remote endpoint information |
ASTERISK-18345: [patch] sips connection dropped by asterisk with a large INVITE |
ASTERISK-18346: MusicOnHold has extra unref which may lead to memory corruption and crash |
ASTERISK-18347: Configure --with-imap fails to handle relative paths |
ASTERISK-18348: Voicemail with IMAP support cannot be compiled under dev-mode |
ASTERISK-18349: Asterisk Crash, with backtrace |
ASTERISK-18350: meetme join causes spontaneous Polycom phone reboot |
ASTERISK-18351: dnsmgr sets port to 0 after a failed DNS lookup |
ASTERISK-18352: add verbose level to logger |
ASTERISK-18353: SIP registration doesn't use round-robin DNS. |
ASTERISK-18354: sqlite crash for realtime action if config_table is not set |
ASTERISK-18355: sqlite realtime_multi_func wrongly assumes commented column exists |
ASTERISK-18356: chan_sip realtime_peer has several memory leaks |
ASTERISK-18357: chan_dahdi does not compile with --enable-dev-mode and gcc 4.6 |
ASTERISK-18358: res_jabber does not compile with --enable-dev-mode and gcc 4.6 |
ASTERISK-18361: http manager getconfig crashes on reading large files/categories (50+ lines) |
ASTERISK-18362: AEL: jump doesn't work as 'jump +123456789;' |
ASTERISK-18388: no meetme recording / file.c:1222 ast_writefile: No such format '' error |
ASTERISK-18389: non-compliant code in chan_sip could be removed for asterisk10 release |
ASTERISK-18390: [patch] New DUNDi cli commands to list cache entries |
ASTERISK-18391: Asterisk crashing in SQLAllocHandle when using ODBC |
ASTERISK-18392: Segmentation fault on Caller ID pattern matching when Caller ID is empty |
ASTERISK-18393: Asterisk 1.8.7.0 Blockers |
ASTERISK-18394: T.38 FAX passthrough does not work |
ASTERISK-18395: Lua applications argument length limitations |
ASTERISK-18396: [patch] - Variables Truncated When Using Realtime Dialplan |
ASTERISK-18399: When using autoload=yes in modules.conf, res_odbc_conf.so complaints about undefined symbol |
ASTERISK-18400: RTCP Receiver Reports are sent for idle RTP sessions |
ASTERISK-18401: Debugging messages generated by 'udptl debug' are incomplete |
ASTERISK-18402: Asterisk accepts a re-INVITE to switch from T.38 back to voice, but does not switch back |
ASTERISK-18403: transfer start ignore digittimout if ! in dialplan |
ASTERISK-18404: out-of-order RTP causes DTMF loss |
ASTERISK-18405: PRI channel becomes unavailable |
ASTERISK-18408: SIP channels stuck |
ASTERISK-18409: [patch] /var/lib/asterisk/moh is no longer created by default if no moh files are selected at build time |
ASTERISK-18410: Defining same SIP device as user/friend and peer |
ASTERISK-18411: Queue members with hints for state_interface get stuck in "In Use" state. |
ASTERISK-18412: iLBC issues during install |
ASTERISK-18413: chan_misdn has a most broken round robin routiune |
ASTERISK-18414: Asterisk DeadLocks After Few Hours of work |
ASTERISK-18415: asterisk 1.8 with 99% CPU usage (Meetme with Moh) |
ASTERISK-18416: [patch] Realtime queue agents unavailable via AMI before a call event. |
ASTERISK-18417: app_alarmreceiver hanging forever in send_tone_burst/ast_waitfor() |
ASTERISK-18418: [branch] Set channel variables when manager originates a call |
ASTERISK-18419: ERROR[1902] utils.c: write() returned error: Broken pipe |
ASTERISK-18420: [regression] Inbound ISDN Overlap dial breaks with Asterisk Version >= 1.8.4.0 |
ASTERISK-18422: Warning in CLI that seems wrong. (Asked to transmit frame type ulaw, while native formats is 0x100 (g729)) |
ASTERISK-18423: Crash in AMI initiated sip show peers |
ASTERISK-18424: TestSuite: Framework: Allow tests to be dependent on build options |
ASTERISK-18425: TestSuite: Framework: Streamline SIPp / pjsua integration with TestClass |
ASTERISK-18426: TestSuite: Framework: Add test execution modes |
ASTERISK-18427: TestSuite: Framework: Add ability for tests to be executed in subsets |
ASTERISK-18428: TestSuite: Framework: Add pre and post test check framework |
ASTERISK-18429: TestSuite: Framework: Add pre / post check for memory usage |
ASTERISK-18430: TestSuite: Framework: Add pre / post check for threads |
ASTERISK-18431: TestSuite: Framework: Add pre / post check for locks |
ASTERISK-18432: TestSuite: Framework: Add pre / post check for active channels |
ASTERISK-18433: TestSuite: Framework: Add pre / post check for active processes |
ASTERISK-18434: TestSuite: Framework: Add pre / post check for SIP dialogs |
ASTERISK-18435: TestSuite: Framework: Add pre / post check for file descriptors |
ASTERISK-18437: TestSuite: Framework: Add post-test data collection and analysis |
ASTERISK-18438: TestSuite: Framework: Add post-test analysis for SIP traffic |
ASTERISK-18439: TestSuite: Framework: Add post-test analysis for Asterisk core dump |
ASTERISK-18440: TestSuite: Framework: Add generic packet capture option for testsuite |
ASTERISK-18441: TestSuite: AstDB: Test Plan Development |
ASTERISK-18442: AstDB: Upgrade warning and instructions |
ASTERISK-18443: Develop a tool to migrate an SQLite3 AstDB back to Berkley DB |
ASTERISK-18444: TestSuite: ConfBridge: Test Plan Development |
ASTERISK-18445: TestSuite: Codecs: SIP test |
ASTERISK-18446: chan_sip rtcachefriends=no loads fullcontact, but doesn't store it, except in astdb |
ASTERISK-18447: Debug manager actions in the CLI |
ASTERISK-18450: Should there be transcoding after attended transfer |
ASTERISK-18453: manager.c: HTTP Manager, fdopen failed: Bad file descriptor! |
ASTERISK-18454: Option for Read to be able to accept # |
ASTERISK-18455: RTCP stats works only when transcoding |
ASTERISK-18457: Crash in timing.c:169 while sound is being played in ConfBridge |
ASTERISK-18479: ast_manager_register_struct attempts to unlock an uninitialized rwlock |
ASTERISK-18480: Linear queue orders real time members alphabetically by their interface. |
ASTERISK-18487: Daily deadlock issue |
ASTERISK-18488: The "pin" parameter of Meetme cmd seems broken. |
ASTERISK-18489: Asterisk 10 Beta - T38 NAT not working |
ASTERISK-18490: [patch] res_rtp_asterisk.c: ast_rtp_read: Bad address cast to IPv4 |
ASTERISK-18491: Deadlock on chan_sip / MASTER_CHANNEL |
ASTERISK-18492: Deadlock in channel.c / chan_sip.c |
ASTERISK-18493: One size sound |
ASTERISK-18494: Whisper disconnects call |
ASTERISK-18495: module unload chan_iax2.so cause mutex errors |
ASTERISK-18496: 1.8.7.0-rc1 breaks dahdi |
ASTERISK-18497: Set default tonezone for SIP devices |
ASTERISK-18499: Asterisk 1.8.8.0 Release Blockers |
ASTERISK-18528: Core Dump on r335064 during system reload. |
ASTERISK-18529: Badly needed function strreplace (issue 0018023) needs to be ported to 1.8 |
ASTERISK-18530: improper use of host LDAP attribute value as ToHost sip client value |
ASTERISK-18531: Asterisk 1.8.6 - MySQL Realtime - "language" variable is useless |
ASTERISK-18532: Asterisk 10.0.0-beta2 Blockers |
ASTERISK-18533: sip channel not closed properly |
ASTERISK-18535: [regression] Asterisk 1.8.7.0-rc1: configure error (libpri related) |
ASTERISK-18541: Crash under heavy load |
ASTERISK-18542: Meetme support have support level 'core' |
ASTERISK-18543: Apparent Deadlock in chan_sip continues, even after repeated efforts. |
ASTERISK-18544: Pure 1.8 from today. It crashes |
ASTERISK-18545: System can crash when using long strings with STRREPLACE() |
ASTERISK-18546: Receive WARNING[28319] res_odbc.c: Limit should be a number, not a boolean: '0'. Disabling ODBC class 'db_name'. |
ASTERISK-18554: CLI 'manager show command challenge' output missing a required field |
ASTERISK-18556: Call parking causes announcment and ringback to caller channel |
ASTERISK-18557: Call Parking parkinghints = yes doesn't work as expected |
ASTERISK-18558: Option for disabling password less logins to voicemail |
ASTERISK-18559: rtptimeout not working per peer |
ASTERISK-18560: Crash while executing macro with CALLERID(num) is empty |
ASTERISK-18562: Call Parking Ringback Happens to Caller Channel, Rather Than Parking Party |
ASTERISK-18565: Voicemail saycid: Play Callers name for external callers if available. |
ASTERISK-18566: G.729 RTP Payload Size |
ASTERISK-18567: app_queue does not add a cdr it should as it is establishing calls |
ASTERISK-18568: Extend the use of Wait to intergrate with res_fax and detect fax/voice |
ASTERISK-18569: Extend the use of Wait to intergrate with res_fax and detect fax/voice |
ASTERISK-18570: Crashes in RTCP handling |
ASTERISK-18571: "core show channel" CLI command blocks the channel during output |
ASTERISK-18572: SIP REGISTER fails if :port appears in the To: header |
ASTERISK-18573: Lock not released |
ASTERISK-18574: SendURL always waits for acknowledgement |
ASTERISK-18575: MySQL is detected on RHEL/CentOS when devel libs aren't installed. |
ASTERISK-18576: ./configure does not pick up missing mysql dev library files |
ASTERISK-18577: National prefix inserted even when caller ID not available |
ASTERISK-18578: Asterisk defaults to s@default in pbx_start if extension is not found |
ASTERISK-18583: 'r' option to Dial() not working as documented |
ASTERISK-18584: SIP Call-ID for B-leg for non-bridged calls |
ASTERISK-18585: Dial() Limit reminds callers at the wrong time than that specified in the L option |
ASTERISK-18586: When Dial() plays warning messages in the LIMIT options, it put the other party into complete silence |
ASTERISK-18587: Enable strictrtp by default |
ASTERISK-18588: Static queue agent penalty not respected (rrmemory) |
ASTERISK-18593: AEL for loops use Macro app and pipe delimiter |
ASTERISK-18602: Voicemail does not read callID from envelope |
ASTERISK-18603: SIP Channels not passing DTMF Tones properly |
ASTERISK-18604: Constant Lockups throughout the day |
ASTERISK-18608: Asterisk 10.0.0-rc1 Blockers |
ASTERISK-18609: When sending fax from Cisco 1751V with t.38, after sending re-INVITE asterisk fully ignore SIP messages from Cisco. |
ASTERISK-18610: ERRORs since changeset 336294 (Fix bad RTP media bridges) |
ASTERISK-18611: RTP packets being repeated and random sequence numbers are being skipped |
ASTERISK-18612: Asterisk logs "Audio is at 5060" |
ASTERISK-18613: Deadlock (SIP not responding anymore) |
ASTERISK-18614: Set Codec for MulticastRTP channel |
ASTERISK-18615: Asterisk Randomly Crashes |
ASTERISK-18616: Call Forward executes callers channel in wrong context |
ASTERISK-18617: ast_srtp_unprotect: SRTP unprotect: authentication failure |
ASTERISK-18618: IAX2 needs to be converted to use bitmask lists for codec selections |
ASTERISK-18619: Segfault when executing 'core show locks' (debug version) |
ASTERISK-18620: Asterisk eating more CPU after upgrade |
ASTERISK-18626: The patch found in r333265 on res_jabber.c breaks authentication to a jabber server. |
ASTERISK-18627: Music on Hold does not play mp3s until after res_musiconhold.so is reloaded manually. |
ASTERISK-18632: Limit PRI channel re-selection to same group in span |
ASTERISK-18634: Create VM_INFO() dialplan function to gather information about a mailbox |
ASTERISK-18635: CallerID is Reverse name vs cid |
ASTERISK-18636: just testing to make a private issue |
ASTERISK-18637: 'Maxforwards' appears twice in a 'SipShowPeer' AMI action response |
ASTERISK-18638: Crash when using chan_unistim |
ASTERISK-18639: Multiple Bridge Events Triggered Upon DTMF Key-Presses |
ASTERISK-18640: ${SIP_HEADER(Subject)} does not get anything |
ASTERISK-18641: T.38 Passthrough Broken |
ASTERISK-18642: asterisk-10.0.0-beta2 , can not run make menuselect |
ASTERISK-18643: [patch] Allow sip devices to have externaddr setting |
ASTERISK-18644: [branch] Add support for early media in AMI action originate |
ASTERISK-18645: There is no difference in queue log between adding member as paused and unpaused. |
ASTERISK-18646: App Dial using Option F: Proceed with dialplan execution at the next priority in the current extension if the source channel hangs up. |
ASTERISK-18647: Can't open XML documentation |
ASTERISK-18648: DAHDI channel causes Asterisk to segfault crash due to unhandled ast_read() NULL return |
ASTERISK-18649: SIP-Use-Reason-Header field badly formatted in the SIPShowPeer AMI call response |
ASTERISK-18650: Asterisk hangs after failed directed call pickup attempt, logs show "Fixup failed on channel SIP/xxx, strange things may happen." |
ASTERISK-18651: Compile failure Debian/sparc64 |
ASTERISK-18652: Asterisk Doesn't Release RTP ports as it should. |
ASTERISK-18653: Parking Slot Number Not Being Cleared |
ASTERISK-18659: If connection address in SDP content equals "sent-by" address of the Via header, then send RTP media to same address as SIP responses |
ASTERISK-18660: ooh323 does not compile in latest 1.8 |
ASTERISK-18661: CLONE - chan_gtalk only receives calls from Gtalk Android |
ASTERISK-18662: Member penalty ignored because wrong queue membercount |
ASTERISK-18663: BLF Subscriptions Causes SIP Deadlock |
ASTERISK-18669: SIP Peer Name case not updating on sip reload |
ASTERISK-18670: documentation for STAT function is a little sparse |
ASTERISK-18671: a new Invite after 5 mins |
ASTERISK-18672: Busylevel doesn't appear to limit calls |
ASTERISK-18673: Large number of active sip dialogs INVITE in the output "sip show channels". |
ASTERISK-18674: CLONE - Flooding with chan_sip.c:6047 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x |
ASTERISK-18675: SendFax T.38 don't work |
ASTERISK-18676: Update thirdparty mISDN from v1.1.9.1 to v1.1.9.2. |
ASTERISK-18677: Update mkrelease and prep_tarball scripts to pull pre-exported documentation |
ASTERISK-18678: Option 'g([[context^]exten^]priority)' for Dial Application |
ASTERISK-18679: Wrong autopause behavior |
ASTERISK-18680: CHANGES and/or UPGRADE.txt files need updating to reflect codec support outside chan_sip |
ASTERISK-18682: Voicemail video "crash" when key is pressed |
ASTERISK-18683: Update wikibot to export Asterisk 10 command reference |
ASTERISK-18684: Export PDF and HTML files for Asterisk 10 from the Asterisk wiki |
ASTERISK-18685: Half Attended Transfers ("Blind Transfers") Fail for No Apparent Reason |
ASTERISK-18687: CLONE - [regression] Asterisk 1.8.7.0-rc1: configure error (libpri related) |
ASTERISK-18688: Reloads with large dialplan cause peers to go lagged |
ASTERISK-18689: Error in documentaion of Application_AGI |
ASTERISK-18690: Unable to enable a disabled module |
ASTERISK-18691: messages WARNING[XXXX] features.c: Failed to play transfer sound! and next the agent goes to log out |
ASTERISK-18692: T38 asterisk Answer 488 |
ASTERISK-18693: "rtp set debug ip" not work |
ASTERISK-18694: Chan_Unistim.c error during compilation |
ASTERISK-18695: CLONE - Asterisk Crash when Realtime LDAP extensions not found |
ASTERISK-18696: peer_iphash_cb empty address on sip reload |
ASTERISK-18697: [minivm] Crash in MinivmNotify |
ASTERISK-18698: Asterisk crashes withou error |
ASTERISK-18699: CDR record not updated when using func_callerid |
ASTERISK-18700: chan_sip.c and tcptls.c - possible double close of file descriptor |
ASTERISK-18701: Sorting of core show channels verbose |
ASTERISK-18702: Improvement of overlap dialling handling in chan_sip |
ASTERISK-18703: NO acceptance of SDP packets with set i= field |
ASTERISK-18704: Asterisk deadlock |
ASTERISK-18705: Asterisk Support of SipConnect 1.1 |
ASTERISK-18706: UDPTL fail while using directmedia |
ASTERISK-18707: QueueSummary event returns incorrect LoggedIn value |
ASTERISK-18708: func_curl hangs channel under load |
ASTERISK-18709: lua socket.http crashes asterisk |
ASTERISK-18710: Alarms not properly set on PRI trunks at startup |
ASTERISK-18711: Advance Call Routing Capability. |
ASTERISK-18712: [patch] Advance Call Routing Capability |
ASTERISK-18713: Autoservice thread is orphaned in a blind transfer during callparking |
ASTERISK-18714: Outgoing calls fail again with Google Voice |
ASTERISK-18715: Allow dialplan to know in advance about estimated wait time for a caller before sending him in a Queue |
ASTERISK-18716: file.c:1352 waitstream_core: Unexpected control subclass '32' after upgrading to asterisk 10.0.0-beta2 |
ASTERISK-18717: Improve error message for app_confbridge |
ASTERISK-18719: res_jabber segfault when using function JABBER_RECEIVE with no message (as when receiving buddy typing notifications) |
ASTERISK-18720: chan_sip stops frquently working (deadlock) |
ASTERISK-18721: 603 decline is busy not circuit-busy |
ASTERISK-18722: ast_expr2 reports "op_times: overflow" on some calculations, though the number is calculated correctly. |
ASTERISK-18723: HANGUP agi message does not show up properly in "agi set debug on" output |
ASTERISK-18724: crash in __ao2_ref_debug |
ASTERISK-18725: Several patches related to the internal editline libraries |
ASTERISK-18726: CDR processing appears to hang during channel hangup update |
ASTERISK-18727: Invalid extension protection upon adding queue member |
ASTERISK-18728: Segfault in app_stack.so on Solaris |
ASTERISK-18729: md5secret can not be used with register strings |
ASTERISK-18730: Asterisk crashes and there is asterisk Failed to start PBX :( in logfile |
ASTERISK-18731: [patch] DUNDi weight parameter not processed correctly |
ASTERISK-18732: T38 gateway : CSI / DSR DSI not sent in reply of remote fax CSI / DSR DSI |
ASTERISK-18733: No CDR for AMI redirect |
ASTERISK-18734: Asterisk crash in click-to-call scenario (SIP only) |
ASTERISK-18735: Asterisk spontaneous reboot |
ASTERISK-18736: Do not retransmit DTMF signal from INFO oriented mode SIP channel to RFC2833 oriented SIP channel |
ASTERISK-18737: AGI Park command attempts to double park the call. |
ASTERISK-18738: 1.8.8.0-rc2: chan_h323 no longer built by default |
ASTERISK-18739: Endianness Problem with Playback WAV Audio on an Big Endian Proccessor ( format_wav.c) |
ASTERISK-18740: Deadlock in queues during dialplan reload |
ASTERISK-18742: PRI Span: 1 !! Unknown IE 128 (cs0) |
ASTERISK-18743: Asterisk Crash with host unknown |
ASTERISK-18744: Asterisk doesn't build on OSX |
ASTERISK-18745: http problem when asking for listcommands |
ASTERISK-18746: RFC2833 stops responding |
ASTERISK-18747: Deadlock in chan_sip / event on send mwi / unsubscribe |
ASTERISK-18748: channel ooh323 hangs up calls after about 1 minute |
ASTERISK-18749: (Only) First attempt to put a call on hold fails when using SRTP |
ASTERISK-18750: crash on parking a call |
ASTERISK-18751: parallel build fails when cleantest calls clean |
ASTERISK-18752: Problems of Text Messaging in Asterisk 10 |
ASTERISK-18753: Asterisk crash when using cdr_adaptive_odbc and sql server isn't reacheable |
ASTERISK-18754: Queues ringinuse=yes does not ring busy extension |
ASTERISK-18755: SDP DTMF negotiation issue: fmtp:101 0-16 |
ASTERISK-18756: Asterisk crashed randomly - moh |
ASTERISK-18757: mohmp3 crashes |
ASTERISK-18758: CLONE - Asterisk ignoring sendonly SDP generated from Cisco UCM after generating inactive SDP when a Cisco phone initiates hold |
ASTERISK-18759: Asterisk re-uses stale nonce in edge case |
ASTERISK-18760: Deadlock in SVN 1.8 version 342359 |
ASTERISK-18761: Create a new hint type for voicemail boxes |
ASTERISK-18762: dialplan remove include without arguments crashes asterisk |
ASTERISK-18763: Different behaviour in case of usage of "#include file" instead of direct config pars |
ASTERISK-18764: Asterisk stopped accepting sip registrations, and stopped logging to full. |
ASTERISK-18765: Memory Leak in lock.h |
ASTERISK-18766: Very poor performance when manager.conf enabled=yes |
ASTERISK-18800: Sending callerid(name) on PRI may cause call rejection |
ASTERISK-18801: Menuselect Easter Egg (Motherships) |
ASTERISK-18802: Adds barriers to the menuselect easter egg |
ASTERISK-18803: [patch] ast_indicate(chan, -1) don't stop playing tones |
ASTERISK-18804: WaitExten(...,m(MOH)) doesn't play the correct audio when Set(CHANNEL(musicclass)=...) is used |
ASTERISK-18805: Remote crash vulnerability in chan_sip when automon in features.conf is enabled |
ASTERISK-18806: Asterisk stops responding to SIP requests after DOS attack |
ASTERISK-18807: [patch] pbx.c silently allows duplicate labels for the same extension, and shouldn't. Suggested [patch] included! |
ASTERISK-18809: pbx_config.c assumes [macro-stdexten] |
ASTERISK-18810: Function QUEUE_MEMBER broken after module reload |
ASTERISK-18811: Dialplan is not processed after AGI script when dst channel hangup first |
ASTERISK-18812: Wrong CallerID |
ASTERISK-18813: Fix misleading gcc warning messages |
ASTERISK-18823: TestSuite: Fix the Asterisk wrapper class to better manage an instance of Asterisk |
ASTERISK-18824: CLONE - TestSuite: Fix the Asterisk wrapper class to better manage an instance of Asterisk |
ASTERISK-18826: Blind Transfer failure |
ASTERISK-18827: iax2 peer/trunk unreachable |
ASTERISK-18828: CEL RADIUS garbage in attribute values |
ASTERISK-18829: ConfBridge deadlocks waiting endlessly for a condition to be signalled inside bridge_channel_join_multithreaded |
ASTERISK-18833: CLONE -Fix misleading gcc warning messages |
ASTERISK-18835: res_monitor causing deadlock with no calls coming through |
ASTERISK-18836: Crash caused by destruction of libc memory pool by destruction of internal asterisk structures: CDR variables. |
ASTERISK-18837: empty Sender-Adress if use TCP-Protocoll for SIP-Messages |
ASTERISK-18838: app_voicemail [general] variables zonetag and locale are not set on mailbox until after reload |
ASTERISK-18839: [patch] missing unlock in sip_send_mwi_to_peer causes deadlock |
ASTERISK-18840: double free or corruption crash with musiconhold |
ASTERISK-18841: Call progress does not work with analog DAHDI cards |
ASTERISK-18842: text config files are treated as unchanged when changing and reloading them quickly one after another |
ASTERISK-18843: moh/loop mp3 files (mpg123) stop playing after core reload when using res_timing_dahdi |
ASTERISK-18844: Extension state callback needs to happen when callback is removed. |
ASTERISK-18845: chan_gtalk only receives calls from Gtalk Android but not viceversa |
ASTERISK-18846: Asterisk crash on issuing Asterisk-CLI 'reload' command multiple times |
ASTERISK-18847: Asterisk 10.0.0 Release Blockers |
ASTERISK-18848: Moduleinfo section in app_macro.c is not terminated properly |
ASTERISK-18849: Add support for video codec media attributes |
ASTERISK-18852: mxml puts quotes inside multiline opaque data |
ASTERISK-18857: misspelling in main/pbx.c |
ASTERISK-18859: Unable to place calls from some SIP phones ("Multiple audio streams are not supported") |
ASTERISK-18862: Change default for sip.conf 'nat' setting |
ASTERISK-18863: Call hang-up when issuing "mixmonitor start" just after "bridge" through AMI |
ASTERISK-18867: Incorrect Password in jabber.conf leads to memory leak |
ASTERISK-18868: sig_pri.c references an incorrect variable inside restart event handler |
ASTERISK-18871: Problem by enabling simultaneous two spans in MFC R2 |
ASTERISK-18873: MEETME_RECORDINGFILE only read when realtime meetme conference is first loaded from the database |
ASTERISK-18875: CDR logs doesn't take care of CDR(dest) variable |
ASTERISK-18876: When g729 codec is configured with packetization time larger than 120 |
ASTERISK-18879: Memory leak inside cel_pgsql |
ASTERISK-18880: CLONE -Memory leak inside cel_pgsql |
ASTERISK-18882: Asterisk lock during production |
ASTERISK-18883: Asterisk TestSuite - test SIP/realtime_sipregs seg faults on exit |
ASTERISK-18885: confbridge, video hangs, Exceptionally long queue length queuing to Bridge |
ASTERISK-18886: database show fails for FOO/BAR/BAZ with sqlite where BDB succeeds |
ASTERISK-18887: Asterisk doesn't respect the codec order - alaw always first in realtime. |
ASTERISK-18889: SRTP packet corruption with SRTCP packet contents |
ASTERISK-18892: CLONE -Asterisk doesn't respect the codec order - alaw always first in realtime. |
ASTERISK-18895: ConfBridge application does not read sound_only_one config variable |
ASTERISK-18897: [patch] SIP Notify Request without "Voice-Message" in the body is not accepted. |
ASTERISK-18899: Erroneous ISDN 44 Rejection Hangup() bug |
ASTERISK-18901: 1.4 app_read.c drops user out during read with no warning |
ASTERISK-18903: Asterisk 10 RC2 Drops To Address from SIP Message |
ASTERISK-18904: Parking timeout does not go to parkedcallstimeout |
ASTERISK-18906: Add manager event on out of call message from SIP |
ASTERISK-18907: AMI crashes on certain commands |
ASTERISK-18908: Meetme - Use voicemail "greet" soundfile for user announce |
ASTERISK-18909: Infinite loop in dialplan pattern parsing |
ASTERISK-18911: When you turn on "sip set debug peer blah" it enables all sip debugging prompts for all SIP channels |
ASTERISK-18912: Realtime MOH with caching plays a new song for every new hold within a call |
ASTERISK-18913: segfault in cdr_adaptive_odbc.c when database connection is interrupted |
ASTERISK-18914: NAT option doesn't show in sip peer's settings |
ASTERISK-18915: Crash on duplicate free in chan_iax2 scheduler |
ASTERISK-18916: Asterisk 10 RC2 Returns 484 Address Incomplete After latest SIP Message fix deployed |
ASTERISK-18917: Asterisk 10 Incorrectly Formats From Header |
ASTERISK-18918: Macro Exit via "Goto()" function, add support for text/named extensions |
ASTERISK-18919: SIP MESSAGE body is not used verbatim |
ASTERISK-18920: Silence after attended transfer on ring |
ASTERISK-18921: sendfax_exec clears LOCALSTATIONID before sending fax |
ASTERISK-18922: crash when you call 'core show channel <channel's name>' |
ASTERISK-18923: res_fax_spandsp usage counter is wrong |
ASTERISK-18924: Linksys devices SIP INFO messages - dtmf-relay signal value uses 0-9, #, *, a-d. Asterisk looking for 0-9, #, *, A-D |
ASTERISK-18925: Asterisk sends "183 Ringing" in sipfrag bodies |
ASTERISK-18926: 10.0.0-rc2 compiles, but chan_sip.so and chan_iax2.so can`t be loaded after compile |
ASTERISK-18927: CLIP India |
ASTERISK-18928: Implicit Assumption About Dynamic Features |
ASTERISK-18929: main/asterisk.c compile error on OpenBSD |
ASTERISK-18930: Asterisk stops responding to SIP devices if it loses Internet Access (DNS) |
ASTERISK-18937: CLONE - Read factory 0xb6d0acb8 was pretty quick last time, waiting for them |
ASTERISK-18938: Park() does auto-answer midway |
ASTERISK-18939: CLONE - [regression] LIMIT_CONNECT_FILE does not get played to caller when using dial() app |
ASTERISK-18940: Channel SIP do not get answer |
ASTERISK-18941: Update Asterisk versions Wiki page with feature-freeze dates |
ASTERISK-18943: Include iLBC source code for distribution with Asterisk |
ASTERISK-18945: Review and refine the 'ignorebusy' option in app_queue |
ASTERISK-18947: Document LINKKEDID_END event |
ASTERISK-18948: core show channels randomly shows IP instead of IAX account |
ASTERISK-18949: Segmentation fault in chan_sip.c (SIP/TLS SRTP configuration) |
ASTERISK-18950: weak (linker) attribute handling for MAC/OS in optional_api.h breaks x86 executable (seg fault) |
ASTERISK-18951: [regression] T.38 pass through produce 100% CPU usage spike |
ASTERISK-18953: Sometimes bridge action fails |
ASTERISK-18955: Voicemail message recording from IAX source sped up and jittery |
ASTERISK-18956: autocreatepeer enhancement (add prefix option for safer peers) |
ASTERISK-18957: asterisk-core-sounds-ru: LICENSE file missing |
ASTERISK-18958: Asterisk Manager incorrectly sets a ChannelID to be global in highly repeatable circumstances |
ASTERISK-18959: astdb2sqlite3 fails to run if it is missing from PATH |
ASTERISK-18961: make fails on cross-compiling for ARM (armVFP) |
ASTERISK-18962: The in CLI documentation for SayNumber is wrong |
ASTERISK-18963: cel_sqlite3_custom creates table |
ASTERISK-18964: Stuttering jittery audio after MOH |
ASTERISK-18966: Consistent AMI error causing Channel variable setting to create global variables |
ASTERISK-18967: [SIP] nonceCaching |
ASTERISK-18968: CALLERID(num) do not work if the fromuser is set ? |
ASTERISK-18969: Followme does not handle inital Connected Line updates. |
ASTERISK-18970: Calls to undefined extensions are not logged into CEL |
ASTERISK-18971: Inbound Gtalk calls fail randomly |
ASTERISK-18973: Asterisk core dumps in video |
ASTERISK-18974: trunk: unable to run unit tests |
ASTERISK-18975: Manager Redirect action on bridged channel pair causes intermittent hangup on second channel |
ASTERISK-18976: pbx_lua and confbridge menu dialplan_exec() do not work together |
ASTERISK-18977: play announcement between music-on-hold files |
ASTERISK-18978: Australia Accented audio files for the conference bridge rewrite. |
ASTERISK-18979: Segmentation fault in scheduled event - send_provisional_keepalive_full |
ASTERISK-18986: Segmentation fault when cdr_mysql.conf file contains errors |
ASTERISK-18987: Alcatel workaround broke EARLY MEDIA for BeroFix |
ASTERISK-18988: Confbridge ghost channels, segmentation fault under high load |
ASTERISK-18989: Dropout in Moh. Seems to be a res_timing_timerfd issue |
ASTERISK-18990: After upgrade from 1.6 to 1.8 one side audio in SPA941 |
ASTERISK-18991: CLONE - Channel SIP do not get answer |
ASTERISK-18992: Asterisk From and To fields setup for SIP out of dialog MESSAGE method |
ASTERISK-18993: CLONE - Flooding with chan_sip.c:6047 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x |
ASTERISK-18994: playback of file format without seeking is broken |
ASTERISK-18995: Support for OGG/Speex file format |
ASTERISK-18996: Faulty SIP session timer handling |
ASTERISK-18997: Segfault in res_config_odbc/res_odbc when using realtime peers (sipregs) |
ASTERISK-18999: Connected Line updates need a disable option on a per SIP device / trunk basis. |